Files
asterisk/apps
Mark Michelson 7122ee6adb Merged revisions 158066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines

Merged revisions 158053 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines

Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.

(closes issue #13867)
Reported by: still_nsk
Patches:
      13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage


........

................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-20 17:40:20 +00:00
..
2008-02-09 11:27:10 +00:00
2008-02-09 11:27:10 +00:00
2008-02-09 11:27:10 +00:00
2008-02-09 11:27:10 +00:00
2007-11-22 01:39:06 +00:00
2007-11-22 04:37:08 +00:00
2008-04-25 20:20:10 +00:00